This is
intended as a general overview of the technical aspects of digital television
distribution. It covers satellite, terrestrial, cable and internet
distribution of audiovisual information.

To
understand the system as a whole, there are quite a few concepts to cover, and
that requires the use of technical terms: these will be explained as
straightforwardly as possible. These fall into some basic categories:

digital
representation

analogue to
digital conversion

data compression

transmission and error correction

reception and storage

Concept 1: digital representation

Digital, at
the most basic level, is the use of just two values. Either the value is 0 or
1. This is unlike most real-world things that can take a wide range of
values.

For many
millennia human civilisation was quite able to get along using ten digits, no
doubt inspired by the collection of fingers and thumbs found to hand. The
Romans used letters for numbers (III for three, VII for seven), the ancient
Egyptians sections of an eye-symbol. It was only several hundred years ago
that 'nothing' gained the familiar ring symbol. This naught gave the ability
of just the ten basic symbols to represent any number by using the position to
denote tens, hundreds and so on.

For humans,
there is little to be gained by using binary. Although it is quite easy to
understand, it is of little everyday use. However, the concept provides
computers with their awesome calculation and storage powers.

Just as the
number 23 actually means 'two lots of ten' plus 'three lots of one', and 15
means 'one lot of ten' plus 'five lots of one', in binary each column makes
lots of (from right to left) 1, 2, 4, 8, 16, 32, 64. Each column value being
twice that to the right, so:

This would
be of limited interest, but making number so simple allows very powerful arrays
of transistors to process the numbers. In the earliest days this was just four
bits at a time (0 to 15), then eight bits (0 to 255), later sixteen bits (0 to
65,535), then 32 bits (0 to 4,294,967,295) and now 64 bits (0 to
18,446,744,073,709,551,615).

The
representation of data in binary form is therefore desirable as it allows high
power, reliable, computers to perform actions that are truly impossible
otherwise. This is because, it turns out, it is much more practicable and cost
effective to make something very simple run very fast.

More than
just counting numbers can be stored using binary digits: they can be used for
other kinds of data. In the 'ASCII' standard, the capital letter A is stored
as 01000001.

Concept 2: analogue to digital conversion

The above
examples have all used positive whole numbers (known as integers), but the real
world is not always like that. Whilst there are plenty of things we can count
(sheep, beans, lamb chops, tins of beans) there are many that we cannot:
temperature, distance, weight or brightness.

If you got
a group of people together and measured their heights you would find two
things. First that you would have a wide range of values, and secondly that
none of them would be exactly a whole number, even if you measured in, say
millimetres. The latter factor would be down to two elements: how carefully
you worked out the value and how accurate your measuring equipment is.

You might
decide to write each value down in millimetres, rounding up or down using a
laser measure. Making this kind of decision turns the analogue values anyone
can be any height into counting values. This process is known as
quantization.

The process
of turning an analogue values is at the heart of the first process used for
digital audiovisual processing: analogue to digital conversion (ADC).

The next
element to add is time. By setting a fast and accurate timer, we can use the
ADC process to produce a stream of values. A simple form of this takes a mono
soundsignal and, 44,000 times a second, makes a value from the current signal
level.

By storing
this data and then using a reverse process (DAC) the original sound is
recreated, almost perfectly. If you have ever listened to a compact disc (CD),
you will be familiar with how well this system works.

There are
limitations only frequencies up to half of the 'sample rate' can be coded
this way.

Encoding by
time ('temporal encoding') is not the only option. A digitalpicture is also
using quantized values to represent the picture elements (pixels) that were
analogue in the real world. In this digital system the values represent red,
green and blue levels in a matrix.

It is also
possible, therefore, to digitize a moving image too. This involves taking
'samples' of a 'digital still' many times a second. This is usually 24 (for
movies), 25 (UK and the EU) or 30 (USA) times a second.

Concept 3: data compression

However,
this generates an awful lot of data: a standard definition television picture
(720x576) at 25 frames per second (25fps) with 24 bits per pixel (that is 8
bits per colour), plus the stereo audio generates:

(720x576x24x25)+(44000x16x2)
bits per second. 248832000+1408000=250240000 bits per second

By
convention, we call 1024 bits one kilobit, and 1024 kilobits one megabit.
Using this example we can see that we would need to transfer 238.6 megabits per
second for a digital TV picture. As this is about thirty times the fastest
broadband connection: this is an impracticable amount of data.

To save
space, we need to compress this data. There are two forms of data compression:
lossless and lossy.

Lossless
compression takes the original data and applies one or more systems of
mathematical analysis to it and (hopefully) spits out less data that can be
then stored. If that stored data is put through the reverse process, the exact
original data is re-created, bit for bit.

This
principle is used by file format such as ZIP, RAR, and SIT that are used to
transfer big files between desktop computers.

However,
there is a small down-side to this type of compression: it is impossible to
guarantee the level of compression achieved it all depends on the source data.
Sometimes you may get a almost no data output, and sometimes you get as much as
you started with. However you can attempt to compress and decompress any type
of data using lossless compression, the program algorithms do not need to know
anything about what the data represents.

If the data
is to be broadcast (or, say, streamed on-line) then there is a need to ensure
that the amount of data is always reduced, so the compressed data can be
transmitted in real time using the available bandwidth.

This calls
for the use for the second type of data compression, called 'lossy'
compression.

Lossy
compression techniques are not general-purpose. They rely on knowing two
things the form of the data that is represented and a little about the target
device for the data: human beings.

For
example, the retina of the human eye has 'rods' and 'cones' packed together.
The 'cones', located in the centre allow us to perceive three colours: red,
green and blue. The 'rods' are away from the centre and react accurately to
many light levels, but only in monochrome. The human brain takes the
monochrome, red, green and blue elements and combines them into full-colour
pictures.

Knowing
this about the human eye provides the simplest form of lossy compression. The
original image is converted from Red, Green, Blue format into three
corresponding values: the hue, saturation and lightness. The first is the
colour, the second the amount of that colour and the final the brightness.

This means
that we can now dispose of some of this data because we humans will still
perceive that the image is the same as demonstrates:

The next
stage is to take the three image components (hue, saturation and lightness) and
break them down into chess-boards. From our original image we will have:

720x576 → 90 x 72 = 6,480 chessboards x 1

360x288 → 45 x 36 = 1,620 chessboards x 2 =
3,240

Each of these
9,720 chessboards is an 8x8 matrix of values, ready for compression. There are
several stages:

first the
'average' value for the whole chessboard is calculated

next each value
on the board is recalculated by subtracting it from the average value

then each of
these new values (which could be positive or negative) are divided by a
'compression factor'.

Then the values
are read from the chessboard in a special zig-zag pattern

Finally the
zig-zag values are then 'run length encoded'. Because many of the values from
the zig-zag 'walk' will be zeros, this achieves good data compression.

Only ONE of
the above stages is actually lossy: the division by the compression factor.
All the other forms do not remove any data.

When this
data is eventually used to recreate the image, the higher the compression
factor the less detail there will be in the recreated image. A very large
factor could result in just a single chessboard with the just the 'average
value' in each square. A low factor will have almost all the original detail.

However, it
is awkward to compute the compression factor value: a fixed amount of output
data is needed for transmission. Too much data would not fit in the capacity
for broadcast, but too little data would result in a first a blurry and then
blocky image.

Concept 4: Temporal compression

The next
compression technique has the marvellous name 'temporal compression'. Under
normal circumstances some or all of the one frame of a TV picture will be
identical to the previous one. By comparing consecutive frames and identifying
those parts that have not changed, the compression system can just bypass these
sections. If the picture is mainly static (such as a 'talking head', such as a
newsreader) the only data that needs to be transmitted is the small sections
that have changed.

The only
drawback to such a system is that a frame that is dependent on a previous one
cannot be displayed if the previous was received: the viewer does not want to
wait for several seconds when 'flicking' between TV channels or for the picture
to 'unjam' if there is just a momentary reception break.

There are
many situations where a considerable portion of the picture does not change
between frames, but moves slightly. This is the final stage of the MPEG2
compression system and the most computationally intense. Having identifying
those sections of the picture that have remained static between frames, the
encoder has to identify which parts of the image have moved, and where they
have moved to.

This is a
very complicated task! There is an almost infinite combination of movements
that could happen. For example, a camera of a football match may pan
horizontally, but a camera following a cricket ball's trajectory has many
options.

TV channels
can have scrolling graphics, fades and wipes; material can wobble or shake.
Objects can move around the screen like a tennis ball. And this can all happen
at the same time.

The better
the encoding software is, and the more powerful the hardware the more motion
can be detected. The better the detection is the less data capacity is
required to describe the moving image and the more can be allocated to
accurately reproducing the detail of those sections that have.

You may
wonder how effective this computing is. Using them all in combination will
reduce the initial 238Mb/s (megabits per second) to as low as 2Mb/s, with
higher quality results at 5Mb/s - a compression ration of from 1:50 to 1:120!

Concept 5: Statistical multiplexing and opportunistic data

This effect
can be enhanced by using more techniques! On Freeview, for example, each
transmission multiplex carries either 18Mb/s or 24Mb/s. By dynamically
co-coordinating the 'compression factors' of a number of TV channels together
using 'statistical multiplexing' one or two more channels can be fitted onto
the multiplex.

And if
there is any capacity left at any time, this is allocated to the interactive
text services (for example BBCi) as 'opportunistic data'.

Concept 6: Audio compression

By
comparison the audio data compression is simple!

The
"MP3" encoding of sound in fact refers to "layer III of
MPEG2". This technique uses some mathematical functions called fast Fourier Transforms to convert each small section of
sound into a number of component waveforms. When these waveforms are
recombined, the original sound can be heard.

The audio compression simply
prioritizes the information in the sections of sound that humans can hear, and reduces
or removes sound information that cannot be heard. As this changes from sample
to sample, the compression routines optimize for each one. This produces a constant
stream of bits at a given rate which is included alongside the picture
information in the "multiplex" (see below).

Concept 7: The "transport stream"

It is worth
taking a moment to consider the multiplexing process a little more. As we have
seen above, the video and audio are highly processed and result in a stream of
bits, and there can be many simultaneous audio and videos to be transmitted
together.

The concept
of a multiplex has nothing to do with a large cinema, but is a mathematical
concept. The actual implementation is quite complex, but the concept is not
difficult.

At the
"multiplexing" end of the system, there are a number of "data
pipes" that have audio, video and other forms of data. The "other
forms" can be the "now and next" information, a full Electronic
Programme Guide, subtitles or the text and still images for a MHEG-5 system
(such as BBCi or DigitalTeletext).

The encoder
takes a little data from each "data pipe" in turn. This amount of
data, called a "packet" is the same size for each incoming stream.
Before the packet is sent to be broadcast, it is "addressed" with a
number of the identify the data pipe from where it came.

At the
receiver, these packets are received in turn. Whilst it is perfectly possible
to decode all the original data pipes, this is not normally required as the
user will normally only be able to view one video and listen to one audio
channel at a time.

This
"demultiplexing" process therefore allows most of the data to be discarded
by the receiver, with only one selected video, one selected audio and one
selected text being used by the rest of the receiver's circuitry.

In
practice, the receiver will also demultiplex and store information that comes
from a number of special "data pipes" provided by the broadcaster.
This will include EPG information, and a directory of the services included in
the broadcast.

For
example, this includes the Network Information Table (NIT) that lists the names
of the channels provided, and the pipe identifiers for the video (VPID) and
audio (APID) for each. This type of information is provided on a constant
loop as it is required when a tuner is scanning for channels during set-up, and
allows for the allocation of the "logical channel numbers" - the
numbers you type into the remote control to view the channel.

Channels
persist in the NIT when they are off-air, allowing channels that broadcast part
time to still be discovered. Radio stations simply have no VPID, with radio
and part-time channels relying on an automatically started text service to
provide some vision.

Just a
final note, the term "statistical multiplexing" refers to the
multiplexer. In contrast to "time division multiplexing" where each
of the incoming data pipes are processed in a "round robin" fashion,
each in turn, the "statistical multiplex" processes each pipe in
turn, but allows "extra goes" for those with the most, or most
critical data: priority is for video and audio, with the text and EPG services
being the least important.

Concept 8: Transmission and error correction

Following
all the processes above, we have a single data stream. There are three main
ways this is broadcast:

All three
of these systems are somewhat different, but they all have one thing in common:
they used to be used to broadcast analogue television. This means that they
are one-to-many, synchronous and unidirectional.

This
differs considerably from most digital computer systems, which are usually
one-to-one (either client-server or peer-to-peer), bi-directional and (usually)
asynchronous. It is for this reason that it has been quite hard to provide
TV services on the internet.

The digital
TV transmission system, COFDM (Coded Orthogonal Frequency Division Multiplexing)
assumes that the path between the transmitter and the receiver will be less
than perfect, and uses a number of further techniques.

The first
is "forward error correction". This is vital because the
transmissions are one-way, not allowing the receiver to ask for corrupt data to
be resent. The most simple way of providing FEC, is to just broadcast every
bit twice. As inefficient as this may sound, this is almost what is actually
done. Using a number of mathematical techniques, this can be reduced slightly,
and is often sent NEARLY twice. The FEC system used DVB-T, DVB-S and DVB-CS
(terrestrial, satellite, cable) is usually quoted as "5/6" or
"3/4", meaning the data is sent one and five-sixths times or one and
three-quarters times.

Concept 9: COFDM

The next
system used is the COFDM itself.

Having
added the FEC to the multiplex data, the COFDM transmitter now takes this and
splits in into 'sub carriers' which are then carried within the analogue
transmission space. The number of sub carriers is 2x2=4 (which gives us Quad
Shift Phase Key), 4x4=16 QAM (quadrature amplitude modulation) or 8x8=64 QAM.
Newer standards such as ATSC (in the US), DVB-S2 and DVB-T2 also use 16x16=256
QAM.

The more
sub carriers that are used, the more data can be carried by the transmission.
However, increasing the number of carriers means that they are all "spaced
closer together", making them more prone to interfering with each other.
In practice, the system used called "phased key shifting" can
compensate for the closeness problem by transmitting them at higher power.

To deal
with the potential interference, the sub carriers do not all broadcast at
once. For much of the time they are unused. The effect of this is that
external interference from analoguetransmitters, other digital transmitters or
anywhere else will cause an error that the FEC encoding can correct. The
amount of time each subcategory is not transmitting is called the "guard
interval".

Thus, more
sub carriers provides more data capacity, as does lowering the guard interval.
But doing these reduces the reliability of the service.

Once useful
feature of this system is that the information decoded from the multiplex can
be stored on a local hard disk drive. It can then, at any time later finish
the decoding process to be replayed as video and audio.

As storage
is the most basic of computer processes (as no computationally complex encoding
is needed) the cost of digitalvideo recorders (also known as Personal Video
Recorder, PVR) is very low. In addition, as the relevant part of the digital
broadcast is stored, replay on these devices is a perfect replay. This
compares favourably to analogue recordings on clumsy video tape which are
imperfect to start with and decay immediately.

Brian, and I am pleased to find that I am getting uninterrupted tv on all channels in my bedroom, on a Phillips (Pace) DTR220 box fed by an old portable set top aerial. Just a 10 inch loop. Brilliant. You have been a great help during the whole process. Thank you so much.

I have a motorhome with a digitalfreeviewaerial. As I move from location to location where can I find information about which direction to point the aerial in and whether it should be vertically aligned or horizontal?

i have great pictures on all freeviewchannels but can't recieve any on HD . do i plug the tv aerial into the aerial socket or do i need a plug for the HD socket? ps my tv is full HD. also i live in northern ireland and should be able to recieve RTE channels but don't.

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